1. Technical Field
The present invention relates in general to the field automated voice processing systems, and more particularly to a method of and system for supporting VoIP media codecs on a voice processing system.
2. Description of the Related Art
Historically, there has been a clear separation between voice communication networks, such as telephone networks, and data networks, such as the Internet. Recently, however, with the advent of Internet telephony, there is then at least a partial merging of the systems. Many calls use a combination of a circuit-switched telephone network and a packet-switched IP network.
The interface between the telephone network and the IP network is a gateway. The gateway performs several functions, including call setup and protocol conversion. As part of the protocol conversion, the gateway converts voice signals from the telephone network to compressed signals appropriate for the IP network. Similarly, the gateway converts compressed signals received from the IP network to voice signals appropriate for the telephone network. The gateway uses codecs (compression/decompression algorithms) to perform the conversions. There are a number of standard codecs used in IP telephony. Many gateways do not support all standard codecs. Accordingly, during the call setup process, the gateways must agree to use the same codec.
Voice processing systems such as interactive voice response (WR) units and voicemail systems have become widespread. Many voice processing systems are connected directly to an IP network rather than to a public switched telephone network. Currently, however, voice processing systems work in “voice mode” rather than IP network “native mode.” The presently existing voice processing systems use recorded voice segments and prompts, which are stored and played as wave files or the like. Presently existing voice mail systems record and playback messages from wave files. Thus, presently existing voice processing systems include an interface that uses codecs and digital signal processors to convert back and forth between voice and network native formats.
The current approach has a number of drawbacks. The amount of computing power needed to perform the compression and decompression in real time can be extremely large, especially the large number of channels are being handled simultaneously by the voice processing system. Such high-powered computing can be provided only by powerful main processors or digital signal processors on adapter cards. For a voice messaging system, the decompression when a voice messages stored in the recompression when a voice messages subsequently retrieved, all in real time, can create voice quality issues, as well as requiring significant processing.